1. Field of the Invention
This invention relates to routing of telephone calls and, more particularly, to connecting of telephone calls over internet protocol networks.
2. Background Information
The traditional telephone network is a switched network that provides users with a dedicated end-to-end circuit for the duration of each call. Circuits are reserved between the originating switch, tandem switches (if any), and the terminating switch based on the called party number to create the end-to-end circuit.
Recently, telephone calls have been transmitted over digital networks using packet switched internet protocol (IP) networks, termed voice over IP (VoIP) transmission. Packet-switched IP networks provide shared, virtual circuit connections between users. Voice information to be transmitted across an IP network is converted into digital data and broken up into multiple, discrete packets. Individual packets may travel over different network paths to reach the final destination where the packets are reassembled in the proper sequence to reconstruct the original voice information. The transmission speed between any two users can change dramatically based on the dynamic number of users sharing the common transmission medium, their bandwidth requirements, the capacity of the transmission medium, and the efficiency of the network routing and design.
VoIP transmission typically costs less than transmission over traditional public switched telephone networks (PSTNs). A disadvantage of VoIP networks is the variability of the quality of the signal received at the destination as determined by changing network conditions. The received signal quality depends on a large number of variable network factors such as packet loss, packet latency, queuing delay, and bandwidth availability. These network factors will vary depending on the volume of network traffic and the location of the destination. The IP network, unlike the traditional public switched network, is not uniformly or predictably suitable for voice quality transmission.
Prior art systems that provide VoIP may monitor the quality of service (QoS) for voice transmissions and select alternate routing for calls when the QoS is determined to be unacceptable. However, QoS is a subjective determination. If the threshold level is too low, some users will have calls routed as VoIP when the QoS is unacceptable to the user. If the threshold level is too high, some users will have calls routed over more expensive lines when VoIP would be acceptable to the user.
The decision to route over IP or alternate routing is often a cost trade-off. The cost of alternate routing generally varies substantially depending on destination. Therefore, a QoS threshold that is suitable for a first destination may be too high for a second destination where the alternate routing is more expensive; the user may be willing to accept a lower QoS because of the higher cost of alternate routing. Similarly, the same QoS threshold could be too low where the alternate routing is less expensive.
The QoS requirement can vary depending on the type of call being transmitted. The QoS required for a teleconference is higher than that required for an automated voice response inquiry. In the case of the automated inquiry, the QoS requirement is different in each direction. The caller will transmit only control tones and a low QoS will be acceptable; the responder will transmit recorded voice and a higher QoS will be appropriate.
As pointed out above, QoS is affected by a large number of network factors. Typically, QoS thresholds are set as thresholds for one or more of the factors that affect quality. However, the factors interact in complex ways. A degradation in one factor can be offset by an enhancement of another factor. Setting thresholds for individual parameters to arrive at an appropriate QoS threshold is difficult. Further, setting thresholds for individual factors disregards the interaction between the factors. The QoS provided when all factors are above the threshold may also be available when one factor is below the threshold if other factors are sufficiently above the threshold.
The International Telecommunications Union (ITU) has issued recommendation G.107, The E-Model, A Computational Model for Use in Transmission Planing (Geneva 1998), that provides a transmission rating model, termed the E-model, for calculating a rating factor, R, based on a large number of terminal and network parameters which are known to impact the subjective perception of end to end voice quality. The recommendation also includes a guide for relating values of R to qualitative measures of voice quality transmission, including Mean Opinion Score (MOS). Higher values of R and MOS correspond to better voice quality and higher QoS. However, computation of R by the full E-model is complex and it is computationally wasteful to use it to compute R values for use in monitoring QoS in real-time.
Accordingly, what is required is a method and apparatus that permits the user to configure the QoS threshold for VoIP connection of calls. The method and apparatus should allow the threshold to be set based on the destination of the call being placed. Further, the method and apparatus should allow the threshold to be set based on an overall QoS desired rather than by setting thresholds for specific transmission parameters.